Mobile terminal and base station in a packet radio services network

ABSTRACT

A mobile terminal for communicating with a base station in a packet radio services network. The terminal has a processor for determining one of a plurality of channels for communication between the mobile terminal and the base station; for digitally coding speech to provide speech information; for assembling speech information into speech packets; and for generating channel allocation requests for a channel in which to send speech packets. A radio transmitter is provided for transmitting the requests and the packets to a base station in the network. A radio receiver receives identities of channels allocated by the base station for the mobile terminal to transmit on. The processor is responsive to each received channel allocation to determine that packets are sent on the allocated channel. 
     In the GPRS since a channel is released when there is no packet to transmit, higher traffic levels can be obtained using the same number of radio channels.

CROSS-REFERENCE TO RELATED APPLICATION

This application claims priority of European Patent Application No.9819136.4, which was filed on Sep. 2, 1998 and European PatentApplication No. 98308584.6, which was filed on Oct. 20, 1998.

BACKGROUND OF THE INVENTION

1. Field of the Invention

This invention relates to packet radio services networks.

2. Description of the Related Art

Standards are being defined for a general packet radio services network(GPRS)

SUMMARY OF THE INVENTION

The invention is based on the recognition that if suitably designed apacket services network could carry speech.

To this end, in accordance with the invention there is provided a mobileterminal for communicating with a base station in a packet radioservices network, said terminal including a processor for determiningone of a plurality of channels for communication between the mobileterminal and the base station; for digitally coding speech to providespeech information; for assembling speech information into speechpackets; and for generating channel allocation requests for a channel inwhich to send speech packets; a radio transmitter for transmitting therequests and the packets to a base station in the network; and a radioreceiver for receiving identities of channels allocated by the basestation for the mobile terminal to transmit on, said processor beingresponsive to each received channel allocation to determine that packetsare sent on the allocated channel.

In the GPRS since a channel is released when there is no packet totransmit, higher traffic levels can be obtained using the same number ofradio channels.

Preferably, if a request to send is not granted, the processor isarranged to discard speech information until a further request isgranted. As speech is highly time sensitive, it is better to discard theinformation than to send the information delayed. The discard producesclipping which, as long as it is not too frequent, is tolerable by theuser.

The processor is preferably arranged so that when a request to send isnot granted, a further request is delayed by a predetermined period.

The delay is preferably increased if successive requests are notgranted.

Following a predetermined maximum delay, the delay is reduced.

The processor is preferably arranged to implement a layered protocol inwhich each packet is given a header in a subnetwork dependentconvergence protocol layer (SNDCP).

Because of the time sensitive nature of speech the header is preferablya RTP/UDP/IP header.

The mobile terminal preferably includes a voice activity detector, andthe processor is preferably responsive to detection of voice activity bythe voice activity detector, to generate a request for a channelallocation in which to send voice packets, and on receipt of a channelidentity, to send the an address header uncompressed on that channelonce and subsequently to send packets with compressed headers which donot contain the destination address on the identified channel, until thevoice activity detector detects no voice activity.

The processor is preferably arranged to construct packets of an equalnumber n of frames, the processor being further arranged to implement alogical link layer protocol (LLC) which adds its own LLC headerinformation comprising a service access point identifier defining speechservice to each packet, and to divide the total LLC plus SNDCP headerinto n parts of equal length and to place one header part before eachframe in the packet. This provides that every frame in the packet hasthe same format and allows a common protection strategy to be applied toeach frame. The header information can be given an error correctingcode. Speech is more error tolerant, however. More important parts ofthe speech information can be coded in order to identify that there isan error, in which case the frame is discarded. Less important parts ofthe speech information can be left unprotected.

Thus, in the physical layer, in each frame, the header and the mostimportant bits speech information are preferably coded using aconvolutional code, and a subset of the important bits of the speechinformation are coded using a cyclic redundancy check.

The invention also extends to a base station including a radio receiverfor receiving requests from mobile stations to send data packets andrequests to send speech packets and operable on a plurality of channelsto receive data packets and speech packets; a processor for reserving apredetermined number of said channels for receiving coded speechpackets, and for allocating nominating a free one of said predeterminednumber responsive to a request channel allocation request in which tosend a speech packet; and a transmitter for transmitting the allocatedchannel to the mobile station.

By dynamically managing the number of channels reserved for speech,optimum service can be given to both speech services and to dataservices given changing respective demands.

The invention also extends to a base station including a radio receiverfor receiving requests from mobile stations to send data packets andrequests to send speech packets and operable on a plurality of channelsto receive data packets and speech packets; a processor for nominatingchannels for a mobile station to send speech packets and for processingpackets in a talk spurt comprising a single destination address headerfollowed by a plurality of speech packets not containing a destinationaddress, for transmission over the network.

The invention further extends to a base station including a radioreceiver for receiving requests from mobile stations to send datapackets and requests to send speech packets and operable on a pluralityof channels to receive data packets and speech packets, a processor forimplementing a protocol which recovers network and transport layerheaders and logical link layer headers for a packet, from equal parts ofeach frame in the packet.

The processor may be operative in each frame to correct errors in theheader and the most subjectively important bits of the speechinformation only.

BRIEF DESCRIPTION OF THE DRAWINGS

One embodiment of the invention will now be described, by way ofexample, with reference to the accompanying drawings, in which:

FIG. 1 is a block diagram of a GPRS mobile terminal and base stationembodying the invention;

FIG. 2 shows schematically the operation of RLC/MAC protocol;

FIG. 3 shows network layer protocol layers;

FIG. 4 shows the SNDCP model operation to support voice;

FIG. 5 shows the format of an LLC-PDU;

FIG. 6 shows the organization of TDMA frames in GPRS;

FIG. 7 shows the GPRS TDMA multiframe structure;

FIG. 8 shows the partition of channels in a GPRS carrying speech;

FIG. 9 shows how source coded bits output from the codec are protectedand

FIG. 10 shows the operation of each layer in the protocol.

DETAILED DESCRIPTION

Referring to the drawings, a mobile terminal 2 has an antenna 4 coupledto a duplexer 6. The duplexer 6 is coupled to a transmitter 8 and areceiver 10. Signals received by the receiver 10 are fed to a processor12. Sound waves of speech are transduced to analog electrical signals bya microphone 14 and the analog signals are converted to digital by theprocessor which may have one or more central processing units (notshown). An analog to digital converter may be a self contained unit 16.The processor processes the digitized speech which is then coded by aparametric codec algorithm, e.g. EFR, to produce speech frames. A codecmay be a self contained unit 18.

A voice activity detector algorithm detects the presence of speechdistinguished from silences. The voice activity detector may be a selfcontained unit 20. When speech is detected, the processor assemblesspeech information output from the codec with network and transportlayer headers into fixed length packets of two frames and sends achannel allocation request.

As may be seen from the block diagram of FIG. 2, if the channelallocation request is refused, a delay is introduced before a newrequest is sent and speech frames to occurring during the delay arediscarded.

A base station 22 has an antenna 24 feeding a duplexer 26. A radioreceiver 28 sends packets received from the mobile terminal 2, to aprocessor 30. Data for transmission to the mobile terminal 2 is sent toa radio transmitter 32 coupled to the duplexer 26.

Network layer protocols, illustrated in FIG. 3, are intended to becapable of operating over services derived from a wide variety ofsubnetworks and data links. GPRS was designed from the outset to supportseveral network layer protocols providing network transparency for theusers of the service. Introduction of new network layer protocols to betransferred over GPRS was to be allowed without any changes to the GPRSnetwork, a function carried out by the subnetwork dependent convergenceprotocol (SNDCP). In addition, SNDCP 40 carries out header and datacompression, and multiplexing of data coming from different sources tobe sent over the LLC layer 42.

IP is used as the network protocol with RTP being used to providesupport for the real time streaming by supplying timestamp informationand packet sequencing. SNDCP currently only provides for TCP/IP andEP(v4) header compression by implementing the RFC1144 compressionalgorithm. However, the SNDCP specifications also allow for additions tothe list of supported compression protocols, according to therequirements of new applications and services. The present systememploys the RTP/UDP/IP protocols which involve an overhead of 40 octets,corresponding to 320 bits.

Using packets of two frames length, it is necessary to support some formof compression for these transport and network layer headers. Indeed, ifthe CS-I channel coding scheme were to be used, the combined RTP/UDP/IPheaders would occupy the entire information payloads of two radioblocks, leaving no space for any speech information or logic linkcontrol (LLC) headers.

A high compression efficiency may be obtained by treating the IP/UDP andRTP headers together rather than separately. Although it is contrary tothe ethos of layered architecture, crossing these protocol layerboundaries is appropriate because the same function is being appliedacross all layers.

There are two main properties of the transmitted packets which are usedto carry out header compression. The first factor-of-two reduction indata rate comes from the observation that half of the bytes in theheaders remain constant over the life of the connection. An obviousexample is the source and destination addresses and ports. Theuncompressed header is sent once, during a connection establishmentphase. These fields are then deleted from the compressed headers thatfollow without any real loss of information.

The remaining compression comes from differential coding on the changingfields to reduce their size. In particular, for RTP header compression,a big gain in efficiency comes from the observation that althoughseveral fields change in every packet, such as the sequence number andthe timestamp, the difference from packet to packet is often constant,and therefore the second-order difference is zero. By making use ofthese properties, the massive combined RTP/UDP/IP header can be reducedto two bytes or three bytes, depending upon whether a header checksum isused. As at least part of the end-to-end link includes at least onemobile propagation path, which is by its very nature subject to error,it would be useful to include the header checksum in the schemeemployed. Although it is not be used for error correction or frameretransmission schemes, it gives an indication that part of the headermay be corrupted and to ignore the timing information for thatparticular packet.

SNDCP also supports data compression by means of the V.42 bis datacompression algorithm. However, as the application layer which sitsabove the SNDCP layer already includes a lossy source coder in the formof a speech codec, there stands little to be gained by applying datacompression by means of entropy coding, as most redundancy in theoriginal information would have been already extracted. In addition,source coding modifies the speech coder bit patterns and makes itdifficult to apply differential channel coding to the speech frameaccording to the subjective importance of the different bit positions.

FIG. 4 shows the SNDCP model operation to support voice. Analysis of theVoice over GPRS delay budget showed that maximum payload efficiency canbe achieved by encapsulating two speech frames into a single networkpacket. Increasing the number of speech frames accommodated by a singlenetwork packet brings about a proportional increase in the packetbuffering delay, thereby increasing the maximum end-to-end delaythreshold of 200 ms.

The SNDCP layer 40 therefore accepts the combined RTP/UDP/IP headers 50and the speech frames through two different service access points.Header compression 52 is carried out, and the resulting header 55segmented into two sections 54, 56 for addition to the two speech frames58, 60 that is encapsulated into that particular packet. This systemallows for the two radio link control (RLC) blocks containing the speechinformation to have exactly the same layout, and therefore use exactlythe same channel coding scheme for both blocks. As the forward errorcorrection is tailored to catering for the properties of a particularspeech coder, it is important to ensure that each bit position with aradio block refers to the same bit position within a speech frame forall transmitted blocks. The first received speech frame belonging to aparticular network packet is forwarded directly to the lower layerwithout waiting for the second frame to arrive.

The Logical Link Control layer 42 operates above the RLC 44 and BSSGP 46layers in the illustrated architecture to provide highly reliablelogical links between a mobile terminal and its serving GPRS supportnode (SGSN). Its main functions are designed towards supporting such areliable link and they include sequence control of LLC frames across alogical link, the detection of transmission, format and operationalerrors on logical link connection, the notification of unrecoverableerrors and flow control.

The operation of the LLC protocol can be better understood by examiningthe format of an LLC-PDU shown in FIG. 5.

As can be seen, the LLC frame header is divided into two main sections,the Address Field 70 and the Control Field 72. In the Address field isthe Service Access Point Identifier (SAPI) 74. This represents a pointat which LLC services can be accessed and provides a means by which theQuality of Service priority can be defined. As ten out of a possiblesixteen different identifiers currently remain vacant in thespecifications, a new SAPI can be defined for voice services,instructing the layers above, namely the SNDCP and the BSSGP about thepriority required by voice packets over data traffic. The conventionalcontrol field contains two sub-fields, represented by N(S) 76 and N(R)78, whose function it is to determine the position of a particular LLCframe within a sequence of frames constituting a single network PDU.However, this function is superfluous within the context of the Voiceover GPRS system there is no segmenting of network-PDUs, as each N-PDUfits exactly into a single LLC-PDU. These fields are therefore to beomitted within the context of transporting real-time voice packets,without any loss of functionality. Each LLC-PDU conventionally ends witha 24-bit long footer containing a frame check sequence. This enables theLLC layer 42 to ensure that the LLC frame is free of errors (within thecapabilities of the CRC check) before passing it on to the network layerat the SGSN for delivery through the backbone network. Should errors befound, it signals for retransmission by means of the RLC layer 44selective repeat request system. However, as repeat request systems arenot used in the present implementation, and as there already exists acyclic redundancy check at the physical layer, the FCS field 90 withinthe LLC-PDU is also omitted without affecting the functionality of thesystem when transporting speech services. Indeed, should this field beretained, it would be merely ignored by the receiving process, as evenif errors were to be detected, the process would still forward thepacket, because as already described, coded speech has an inherentinformation corruption tolerance.

The system therefore accepts the two segments of the SNDC-PDU containingthe two speech frames which belong to the same network packet, and addthe new, 8-bit LLC header containing the SAPI for voice services to thefirst arriving segment so that the two frames in the packet have headersof equal length. This is then forwarded to the RLC 44/MAC 45 layer forimmediate dispatch over the radio interface. When the peer LLC processat the BSS receives the first radio block containing information fromthat particular LLC-frame, it identifies that it contains speechinformation by examining the service access point identifier. Thisinformation instructs the process not to look for further headerinformation after the first 8 bits, and not to expect any frame checksequence. The space freed by removing these fields is used to carry moreuser payload information.

The RLC/MAC (medium access control) layer provides services for thetransfer of upper layer PDUs using a shared medium between multiplemobile stations and the network. This transfer may take place in theform of unacknowledged operation or acknowledged operation, according tothe nature of the service required. As its very name implies, theRLC/MAC protocol actually consists of two separate protocols withdifferent functions. The Radio Link Control Layer defines the proceduresfor segmentation and reassembly of LLC PDUs into RLC/MAC blocks and inRLC acknowledged mode of operation, for the Backward Error Correction(BEG) procedures enabling the selective retransmission of unsuccessfullydelivered RLC/MAC blocks. When operating in the RLC acknowledged mode,the RLC layer preserves the order of higher layer PDUs provided to it.On the other hand, the MAC (Medium Access Control) function definesprocedures that enable multiple mobile stations to share a commontransmission medium which may consist of several physical channels. TheGPRS MAC protocol allows for a single user to use more than one timeslotconcurrently so as to increase throughput. In addition, the sametimeslot may be multiplexed between up to eight users, so as to increasethe number of users operating on a given set of system resources. In theVoice over GPRS system, there is no provision for multislotting, and forthe duration of time for which a channel is occupied by a single usertransmitting speech information, this channel is not multiplexed withany other users. In addition, no use is made of the retransmissionfunctions of the RLC layer

GPRS shares the same radio interface as the GSM circuit-switched voicesystem. This means that each physical frequency channel is divided intoeight traffic channels by means of time-division multiplexing. Eachtimeslot within a TDMA frame can be dynamically allocated to GPRSservices or GSM services according to the relative shifts in demands forthe two services, with those channels allocated to packet data trafficbeing referred to as Packet Data Channels (PDCH)

Each time division multiple access (TDMA) frame 100 lasts for 4.615 msand can accommodate eight PDCHs within its eight timeslots 102. The datato be transmitted by means of the Packet Data Traffic Channels (PDTCH)is segmented into units of 114 bits each, which are then encapsulatedinto radio bursts for insertion into a single TDMA timeslot which lastsfor 576.8 us. This means that each RLC/MAC block consisting of 456 bitsis segmented and interleaved into four consecutive radio bursts.

GPRS multiplexing differs from that found GSM circuit-switched speech inthe way the TDMA frames are organized into multiframes. Whereas GSMsupports 26-frame and 51-frame multiframes, in GPRS the TDMA frames areorganized into 239.980 ms-long multiframes 104 consisting of 52 frames.This organization is divided into 12 radio blocks of four frames each,and four idle frames as shown in FIG. 6.

Referring to FIG. 7, it should be noted that all 52 bursts in thepicture belong to the same timeslot, and consequently to the same PDCH.In GPRS, several logical channels not found in GSM circuit-switchedservices are introduced and accommodated onto the GSM physical channels(PDCH)

Referring to FIG. 8, the Packet Data Traffic Channel, which is used toactually carry the speech information and the Packet Access Grantchannel which is used for channel contention are mapped onto the PDCH bywhat is known as the ‘Master-Slave’ concept. In this system, at leastone PDCH (or timeslot), acting as a master, accommodates user data anddedicated signaling, and packet common control channels that carry allnecessary control signaling for initiating packet transfer. Such controlsignaling refers only to the access bursts on the packet random accesschannel (PRACH) 120. All other PDCHs, acting as slaves are used for userdata transfer and dedicated signaling only.

For GPRS the master channel is capable of bearing both the PRACH and aPDTCH simultaneously by sharing the physical channel between the twological channels by means of a time-division multiplexing mechanism.Usually, the bulk of the physical resources of the Master timeslot areallocated to carrying user data, with one block every t blocks beingdedicated to supporting random access attempts, where t is typically aninteger with value 3, 4 or 5. In this way, whereas seven slave channelsare entirely dedicated to supporting voice and data traffic, the eighthchannel, which is usually time slot zero (TSO) is used both as a trafficchannel, and as a random access medium for all terminals operating atthat particular radio frequency channel.

The present implementation for supporting voice services requires thatthe master channel be left for control signaling only and not be allowedto accept any PDTCHs. The reason for making such a reservation is theextra delay that sharing the master channel would impose on the averageaccess time. If, for example, t was set to three, one radio block out ofevery three available radio blocks in the master channel would bededicated to supporting the PRACH. This means that a mobile terminal hasto wait approximately 3 RLC blocks, equivalent to 55.3 ms until it isallowed to fire the next random access. This extra delay is clearlydesirable for real-time voice services.

The remaining seven slave channels are then divided into those PacketData Traffic Channels dedicated to voice services 122 and thosededicated to data services 124. The system does not allow a PDTCH toshare voice and data services, as the delay requirements are radicallydifferent. By allowing GPRS voice users access to channels dedicated tocarrying voice services, better control can be made by the base stationto ensure that the required Quality of Service over the radio link interms of access delay and speech frame loss rate are met.

The operation of the RLC/MAC protocol is summarized in FIG. 2. If aGPRS-terminal is voice-capable and wishes to initiate a conversationover the GPRS network, it initiates a call-setup procedure. In thisprocess, that the Base Station monitors the current load in the cell ofoperation of the user and determine if it can support another voiceuser. If it can, the base station informs the mobile terminal that ithas been admitted to the network and may initiate transmission of voicepackets. The mobile terminal then goes into an idle mode where it waitsfor an indication from the terminal's voice activity detector (VAD) thata speech activity has been detected and a talkspurt has begun. A randomaccess burst is sent over the PRACH 140, and it waits for a reply fromthe Base Station over the PAGCH 142. If channel resources are available,it indicates to the MS that it has allocated a single channel for thetransmission of the talkspurt. In order not to be any more channelinefficient than the equivalent GSM circuit-switched voice services,multi-slotting is not enabled for the transmission of voice servicesover GPRS. This means that a single talkspurt may be transmitted in asingle timeslot only, and the base station does not allocate more thanthis single PDTCH for this purpose. The mobile terminal then proceeds totransmit all the RLC/MAC blocks belonging to that particular talkspurt144. On the onset of the following silence period, the mobile stationstops transmitting RLC blocks, thereby indicating to the BS that it isreleasing the channel 146. This means that channel contention occurs atthe beginning of each talkspurt only, and once a terminal has acquireduse of a traffic channel, it only relinquishes it when all the LLCframes corresponding to that particular talkspurt have been transmitted.The Base Station then allocates this channel to the pool of channelsavailable for the transmission of voice services. If, however, there areno available PDTCHs allocated for the support of voice services, the BSinforms the MS of the situation 148. The mobile station then enters arandom exponential backoff period, at the completion of which itreattempts to access the channel 150.

As real-time speech is time-sensitive information, all the RLC/MACblocks corresponding to speech frames generated during the backoff, savethe most current one are discarded 152. For every failure to access thechannel, a counter is incremented 154 and used to determine the durationof the backoff period. If however a collision occurs between accessbursts in the same timeframe on the PRACH, the Base Station is not ableto determine which MS initiated the request, and so is unable torespond. The mobile terminal notices that such a collision has occurredby employing a time set to a value slightly longer than the averageresponse time of the base station. Should this value expire, the mobilestation enters the backoff period in the same way as when channel accesshas been denied.

Both data terminals as well as voice terminals implement an exponentialbackoff algorithm, where after an unsuccessful access attempt, a uniformdistributed random number w e[0,2_(n+1)] is drawn, where n is the numberof access attempts. For data terminals, the next random attempt is triedafter a waiting time of w*8.5 ms, while for voice terminals, multipliersof 4.615 ms and 8.5 ms respectively are used.

As real-time conversational voice is a delay-critical service it isimportant that the medium access algorithm is geared towards keeping theaccess delay of the system to a minimum. A uniform distributed randomnumber is selected from the range [0,2_(n+1)−1] The two parameters whichwhen varied can alter the delay performance of the system are themaximum increment allowed, n, and a constant by which the randomvariable is multiplied in order to obtain the waiting time. Experimentshave been carried out with an unbounded exponential backoff, meaningthat no limit was set to the value of the increment counter, and with atime multiplier of one radio block, corresponding to 18.5 ms. Theexperiments however showed that system performance could be improved byfixing n to an upper limit of 5, and reducing the time multiplier to theduration of a single TDMA frame, i.e. 4.615 ms.

The physical RF transmits information bits over the radio path and dealswith issues such as the carrier frequency characteristics and GSM radiochannel structures, the modulation of transmitted waveforms and thetransmitter and receiver characteristics and their respectiveperformance requirements. GPRS shares all these specifications with theGSM speech standards, and as a consequence can make use of much of thecurrent GSM radio infrastructure.

The Physical Link Layer operates above the physical RF layer to providea physical channel between the mobile terminal and the Base StationSubsystem (BSS). One of its main responsibilities is forward errorcorrection coding (FEC), which allows for the detection and correctionof transmitted code words and the indication of uncorrectable codewords. In addition, the physical link layer carries out rectangularinterleaving of radio blocks over four bursts in consecutive TDMAframes, and procedures for detecting physical link congestion.

GPRS currently supports four channel coding schemes, ranging from ahalf-rate convolutional scheme (CS-I) to a scheme which provides for nochannel coding (CS-4)

Schemes CS-2 and CS-3 are punctured versions of the CS-I scheme, givingresulting code rates of approximately 2/3 and 3/4 respectively.

When deciding which of the currently available coding schemes to use toprotect coded speech, it is desirable to examine the speech codingtechniques used to protect speech in circuit-switched GSM systems. As anexample, we prefer to use the new GSM Enhanced Full Rate Coder (EFR).This is an Algebraic Codebook Excited Linear Predictive (ACELP) coder.It is essentially a parametric, rather than a waveform coder. This meansthat the speech is represented by a number of parameters describing,amongst other things, the pitch period of the speech, a number of LPCcoefficients representing the waveform shaping effects that occur in thevocal tract and a description of the excitation that is generated by thespeaker's vocal chords. Though these parameters are generated so as tomaximize the coder efficiency by bringing to a minimum the redundancypresent in the information being transmitted, not all parameters have anequal effect on the perceptible audio quality of the speech. This means,that when in error, some parameters, and consequently bits, cause agreater distortion in speech quality than other bits. This phenomenon isexploited when protecting the source coded bits by means of FEC.

FIG. 9 shows how the 244 source coded bits 160 output from the EW codecare protected to varying degrees according to the subjective importanceof the bits. Type 1 bits are protected using a half-rate coder identicalto that used by the CS-1 scheme, while a subset of 65 bits, includingall the Class 1 a bits are protected by means of a cyclic redundancycheck (CRC). This is used in order to test for channel conditions. Ifthe CRC check on these 65 bits which represent the subjectively mostimportant bits detects an error, then the receiver considers the speechframe as having suffered an unacceptable degradation in audio qualityand consequently drops the frame. In order to maintain a comparablespeech quality with that offered by OSM circuit switched services, it isnecessary to maintain the same level of channel protection on thesebits, because otherwise there would result an increase in the framedropping rate, with a corresponding decrease in the ensuing speechquality. If the current GPRS coding schemes are to be used, the onlyscheme which would provide this level of protection is CS-1. This wouldhowever protect all bits indiscriminately to the same extent withouttaking into consideration the subjective weighting of the source-codedbits. Using the CS-I scheme would leave a 181-bit payload, whichcorresponds to 9.05 kbit/s. However this figure does not take intoconsideration the headers belonging to the RLC/MAC layer, which as canbe seen in FIG. 10, occupy 21 bits. This further reduces the informationpayload to 160 bits, corresponding to a data throughput of 8 kbit/sadding the LLC and SNDCP headers into the information payload, furtherreduces the user data throughput to about 5.6 kbit/s, depending on whatfunctionality is included in the headers of those layers.

A far more attractive solution to implementing channel coding within theVoice over GPRS system is to use a channel coding scheme tailored aroundthe requirements of the particular speech coder being used. This allowsfor the greatest efficiency, by only offering powerful protection to thesubjectively most important bits, while offering different levels ofchannel coding to the remaining bits. This scheme also allows for theheaders belonging to the RLC/MAC LLC and SNDCP layers to be powerfullyprotected. The present solution is to use the CS-I coding scheme toprotect those radio block data payload bit positions which are occupiedby these higher level headers.

Current coding schemes specify the use of a CRC-based Block CheckSequence (BCS) to detect errors within a radio block. If any such errorsare detected, the RLC layer is informed, and retransmission requested,if operating in acknowledged mode. The BCS is also used by the basestation subsystem to monitor channel conditions between the mobilestations and the base station. In all coding schemes except CS-I, thelength of the BCS is sixteen bits and it operates over the entire lengthof the radio block. As retransmission is not used in the present voicesystem, it suffices to be able to detect errors over the section of thedata payload used by the protocol headers only. This allows for areduced-length sequence of 8 bits to be used.

FIG. 10 shows the GPRS system using the enhancements to the protocols asdescribed above. At the SNDCP layer, header compression is carried outon the joint RTP/UDP/IP headers 180 to give a compressed SNDCP header182 which is only 24 bits long. The SNDC-PDU is then reorganized in sucha way that the header is segmented into two sections of differing length184 and 186, so as to allow for an identical payload format in bothtransmitted radio blocks. This means that while 8 bits of the SNDCPheader are retained at the beginning of the packet, the remaining 16header bits are placed towards the middle of the packet.

The SNDCP-packet is then encapsulated into an LLC frame 188 with theaddition of the new, truncated 8-bit LLC header 190. The RLC/MAC blocks192, 194 are formed by segmenting the received LLC packet into two. Inorder to implement the asymmetric buffering described above, the firstproduced speech frame is forwarded down to the RLC/MAC layer, togetherwith 16 bits of combined SNDCP and LLC headers for immediate forwardingover the radio channel. This is followed 20 ms later by the secondsegment of the LLC frame, this time containing the remaining 16 bits ofthe SNDCP header and the contents of the second speech frame. Thissystem ensures that both RLC/MAC blocks have exactly the same layout. Atthe RLC/MAC layer, a further 21 bits of header 196 are added to bothblocks together with a 3 bit USF 198 and a short 8 bit BCS 200. Eachblock is then channel coded using a new coding scheme optimized forspeech, which is referred to as CSS-1 (Coding Scheme-Speech I). Althougheach speech coding implementation uses a different channel codingscheme, it is assumed that the header information is always powerfullyprotected using a half-rate code. This leaves 360 bits 202 as a grossspeech payload, which is used to contain both the speech information aswell as the portion of channel coding for speech only. This translatesto a data throughput of 18 bit/s, which is a considerable increase onthe 5.6 kbit/s available using the current standards, even when takinginto consideration the throughput in the present scheme that has to beused for channel coding.

We claim:
 1. A mobile terminal for communicating with a base station ina packet radio services network, said terminal comprising a processorfor determining one of a plurality of channels for communication betweenthe mobile terminal and the base station; for digitally coding speech toprovide speech information; and for assembling speech information intospeech packets; for generating channel allocation requests in which tosend speech packets; a radio transmitter for transmitting the requestsand the packets to a base station in the network; and a radio receiverfor receiving identities of channels allocated by the base station forthe mobile terminal to transmit on, said processor being responsive toeach received channel allocation to determine that packets are sent onthe allocated channel wherein if a request to send is not granted, theprocessor is arranged to discard speech information until a furtherrequest is granted and the further request is delayed by a predeterminedperiod where the delay is increased if successive requests are notgranted and where following a predetermined maximum delay, the delay isreduced.
 2. A mobile terminal as claimed in claim 1, wherein theprocessor is arranged to implement a layered protocol, and wherein eachpacket is given a network and transport layer header in a subnetworkdependent convergence protocol layer (SNDCP).
 3. A mobile terminal inaccordance with claim 1 further comprising a voice activity detector,wherein the processor is responsive to detection of voice activity bythe voice activity detector, to generate a request for a channelallocation in which to send voice packets, and on receipt of a channelidentity, to send an address header uncompressed on that channel andsubsequently to send packets with compressed headers which do notcontain the destination address on the identified channel, until thevoice activity detector detects no voice activity.
 4. A mobile terminalas claimed in claim 3, wherein the processor is arranged to constructpackets of an equal number n of frames, the processor being furtherarranged to implement a logical link layer protocol (LLC) which adds itsown LLC header information comprising a service access point identifierdefining speech service to each packet, and to divide the total LLC plusan SNDCP header into n parts of equal length and to place one headerpart before each frame in the packet.
 5. A mobile terminal as claimed inclaim 4, wherein in the physical layer, in each frame, the header andthe most important bits speech information are coded using aconvolutional code, and a subset of important bits of the speechinformation are coded using a cyclic redundancy check.